Fusion 3.0 Limitations

Introduction

The UC Template is a modular solution for IP-telephony and Unified Communications. The solution is based on open standards and is very efficient from an operational point of view. The UC Template is software delivered as a service (SaaS). Updates and a water-proof disaster recovery pack (automated and guaranteed back-up, documentation, redundancy) are included. The solution is most scalable and allows you to effectively respond to the changing communication needs of your organisation. The UC Template is a rich IP-PABX solution, designed to maximize the self-management by IT administrators and end-users. This guide will help you to go through the configuration of your UC Template solution.

Intended audience

This guide is intended for certified engineers only.

Disclaimer

This document is confidential. It is provided for informational purposes only and is not guaranteed to be without errors. Some of the functionalities described may be specific to a device, to an application or to a network. Some features may only be available as optional functionality. Escaux has a policy of continuous improvement and therefore reserves the right to modify this document without prior notice.

Goal of this document

The goal of this document is to describe as much limitations as possible. Please note that this is a non-exhaustive list. New limitations will be added if discovered.

Known issues can be found in the release notes

Version specific limitations

General Escaux Fusion 3.0 limitations

FMU

• FMU users using a fusion sop are not able to use short codes or internal extension with a length bigger then 8. In fact this is a limitation in FMU. For more information check the FMU release notes.

SMP

• Webinterface currently only support English language
• Bulk administration via .csv file upload. No provisioning API available.

Redundancy limitations

See Product Specification: Active-Standby and Active-Active redundancy

There is no active-active redundancy for Fax devices.

Queue members and login status are not synchronized in active-active architectures.

Note also that the SOP does not support clean power-off signals from a UPS. When the UPS is drained, the SOP will be subject to shutdown as a result of a power failure.

Cluster limitations

• Managers and their assistants need to be configured on the same SOP
• CallQueuers and their members need to be configured on the same SOP
• HuntGroups and their members need to be configured on the same SOP
• ATA boxes used for physical fax devices in T.38 mode should be configured on the SOP where faxes are coming in (no T.38 on MeshSipTrunks)

Interface card combinations.

• See Offer Generator: ISDN and analogue connectivity options

Networking limitations

• Ethernet network interface configuration
  • A SOP should not be used to bridge, route or firewall between two networks.
    • Only a single physical interface is supported.
    • Only a single logical interface is supported, no VLAN trunking on SOP.
  • The SMP allows to crate a maximum of 4 static routes.
  • The SOP Ethernet interface can only be configured to carry a static IP address.
  • Currently only IPv4 is supported, no IPv6.
  • The SOP Ethernet interface supports up to Gigabit Ethernet
• DHCP Server
  • No MAC reservations possible
  • TFTP server option supported for Polycom, Aastra, Snom, Mitel and Cisco
  • In an active-active configuration, 2n IP addresses are needed for n phones
• NTP server
  • Maximum 3 NTP master servers can be configured
  • Fixed NTP synchronisation timer
• TFTP Server
  • Only for internal use. Customer content is not supported.
  • Phone configuration files are stored and distributed in clear text.
  • No authentication on TFTP server access.
• Mail Server
  • Only used for outgoing email initiated by the SOP, e.g. voicemail to email or emails sent from within callflows.
  • Only capable of communicating with the customer's email server using SMTP port 25.
  • Customer's email relay server should authenticate SOP based upon the SOP IP address. No other authentication mechanisms supported.
  • Mails are always sent in English
    • Voicemail to e-mail
    • Fax to e-mail
• SNMP Agent
  • SNMP access to Standard MIB is supported
  • SNMP access to Escaux Probe MIB is only used by the BMS servers and not documented for external use.
  • SNMP polling only, no SNMP traps are being sent.

Local SOP interface limitations

• Shell access
  • Limited to Service Assurance and basic configuration tools, no full SOP management.
  • Limited to two accounts. One for the SMP owner and one for the End-Customer.
  • Requires password protected SSH access. Telnet access not possible.
  • Requires direct SOP access. Access not possible via SMP.
  • Full Linux command line access is not available nor supported.
  • A terminal of minimum 80 columns and 30 rows is required to fully display the shell.
• File Manager access
  • Two access levels: one for the SMP owner and one for the End-Customer.
  • Only one account for the End-Customer.
  • Web interface: supports to upload, download and erase a single file at a time.
  • Webdav: supports bulk upload and download of files. Bulk file deletion not supported.
• Database Server
  • No customer access to internal databases supported

Telephony features limitations

General telephony limitations

• It is not possible to use an Idle Url with a Polycom VVX phone. In this case you should configure a softkey with a Status Url.
• Because of the limitation above it is not possible to use the same SVE2xxxx resource for a Polycom IP and VVX phone.
• Based upon Asterisk 1.8 with custom extensions.
• SIP and RTP traffic are carried over UDP, not TCP
• Supported codecs: G.711 µ-law & a-law, G.729 Annex A, iLBC
• QoS: All IP packets sent by SOP carry the same DSCP tagging, irrespective of the destination to which they are sent. A distinction is made though between SIP and RTP.
• Personal User Mobility
  • Only supported for Aastra and Polycom
  • Timeout: 3 minutes grace period to be respected for login after logout (or vice versa)
  • PUM only possible between phones of same brand (Aastra or Polycom).
  • PUM on Polycom phones is only possible between the same type of phones (IP or VVX)
• Sounds
  • Default prompts only available in English, Dutch and French
• Phone Zero Config
  • Tool only intended to be used during initial phone deployment. Not to be used as lightweight version of PUM.
  • Phones should be activated the same calendar date as the last boot of the phone.
• Music On Hold
  • 3 default MoH files played in sequence. These songs are Royalty Free.
  • Customer can upload own MoH files but carries responsibility for royalties.
• Connected line update only works when a phone starts ringing or picks up

Reserved extension range

Some extensions can not be used. These extension ranges can be found here

Callflow limitations

• Business codes
  • Can be set on all outgoing calls but only on external incoming calls
  • Sip can re-invite will automatically be deactivated (configurable on user level) if it should be possible to set the business code for incoming calls
  • Can only be set at the start of the call for outgoing calls
  • Needs to be set before transferring the call for incoming calls
• Call Forward
  • The destination of a call forward can be restricted.
    • The restriction is deactivated for the unconditional forward if the option 'use transfer instead of redirect' is activated. If you are uncertain about your settings, please contact your integrator
  • The option to keep the original number is only available on some SIP platforms and only works for CFU (not for CFB, ...)
• Call Waiting Toggle
  • If call waiting is activated, the user can receive two or more simultaneous calls depending on the phone model. If the phone supports it, the call waiting is not limiting this to two simultaneous calls.
• Emergency calls
  • Extensions defined in the internal directory take precedence over the pre-defined emergency numbers
  • Emergency numbers are defined in the range 100-119, optionally preceded by a zero
• Group recorder
  • To access the recording function of group calls prompts, the user will have to enter a pincode. There is only one pincode for the whole system.
• IVR recorder
  • To access the recording function of IVR prompts, the user will have to enter a pincode. There is only one pincode for the whole system.
• German is not supported
• Pickup
  • limited to 30 groups for group pickup
  • no pincode required for pickup (ex: extension pickup)
  • pickup is not supported between 2 people connected to a different SOP
• Pincode changer : pincode length limited to 4 digits, with forbidden pincodes (1234, 4321, 1111, ....)
• Queue
  • Not possible to auto logoff if no answer
  • A call does not go to an agent with a higher penalty when the agent with the lower penalty is idle
  • A call does not continue ringing on the operator phone during periodic announcement. During the periodic announcement, the caller is taken out of the queue and re-queued again afterwards. The caller looses his position in the queue.
  • Not possible to set penalties on permanent queue members
  • Maximum 100 Queue profiles supported
  • Maximum 50 members per queue supported, if more members are needed a scoping must be done to understand the exact use case
  • Queue login/logout are triggered by Intentional Status change
    • The administrator can define on system level (not on user level!) what should happen for each status (=the same for everyone)
    • Status change via the SMP will not trigger the queue login/logout service when the SMP version is prior to 5.3.0.
• Service availability
  • Max 5 schedules (for 7 days)
  • Each schedule is limited to 4 entries per day: typically for (1) opening in the morning, (2) closing for lunch, (3) opening after lunch and (4) closing in the evening.
  • Maximum 10 Service Availability profiles supported
• Status changer
  • Only accessible from internal extension or pre-defined DDI's (has to be in home or mobile number field in the internal directory)
  • There is a timeout before a new status is shown on screen of the IP Phone
• Groups
  • login logout is only supported on call queuers, not on HuntGroups
  • When a call to a group is answered by a group member, the other group members will have a missed call on their phone.
• Conference
  • simultaneous calls/participants?
  • Maximum 20 different conference rooms per SOP
  • Maximum 40 participants per conference room
  • Changing the pincode requires administrator rights
  • The optional announcement which can be played for an entering/leaving participant is played in the language of the entering/leaving participant, even if the other participants have other languages configured
  • There is no possibility to close the conference room
  • There is no possibility to eject people from a conference room
  • There is no possibility to mute yourself in the conference room
• User/Manager
  • Can be a member in maximum 3 dynamic groups
  • The second action is always executed after the first action has completed (even if the first action is, for example, 'play holiday message')
  • There is no busy tone as second action or as fallback action when a user is already on the phone
• ShortNumber
  • Maximum 1000 ShortNumbers supported.
  • A user using a phone that is part of a restriction group can abuse the ShortNumbers to call un-authorized destinations, e.g. if a ShortNumber points to an international mobile number, a user who's phone is restricted to internal calls only can reach the pre-configured international mobile number by dialing the speeddial.
• Callback
  • Callback is automatically cancelled after 1 hour
  • The implementation is based upon polling every minute. As a result a delay up to 1 minute might occur before the call back is triggered.
• When calling an unauthorized number (due to call restrictions), no feedback is given (no audio prompt saying 'you are not allowed to call this number')
• Route group
  • settings are only supported on single sop configurations
  • applies only for User/Manager/Receptionist profiles (not Short numbers and IVR for example)
• Timechecker
  • Timechecker is not executed when it already passed another timechecker of another extension.

Voicemail limitations

• The maximum number of voicemails plus ad-hoc (via net.Desktop) recordings per user is limited to 100.
• The Welcome Message of the voicemail cannot be longer than 60 seconds.
• There is no enforced limitation in the length of a voicemail. Voicemails are stored on the /data partition.
• There is no enforced limitation in the length of an ad-hoc recording. Recordings are stored on the /data partition.
• There is no enforced limitation in the amount of voicemail or recording users. Voicemails and ad-hoc recordings are stored on the /data partition.
• The size of the /data partition is being monitored via pro-active monitoring.
• Voicemail to email: deleting email does not delete the message on the SOP.
• Advanced options in Voicemail menu like forward a message or reply to a message are not supported for Cluster architectures (M7148)
• When using voicemail as 2th action in an IVR/CallQueuer profile, please make sure the extension you choose to leave the voicemail is on the same sop as the IVR/QallQueuer extension.

Phone support limitations

• Polycom Phone Support
  • Single timezone support per TFTP server
  • Single firmware version per model supported
  • QoS: All IP packets sent by IP phones carry the same DSCP tagging, irrespective of the location of the phone.
  • The corporate directory does live lookups on the SOP. The contacts include SMP contacts and CRM contacts (combined total: 30.000 contacts). The number of results that is shown on the screen after a search query is phone dependent.
  • When using PUM, the local directory (and speeddials) are stored on the phone itself and don't follow when logging in on a different phone
  • Speeddials and the local directory can only be provisioned via the phone itself
  • When a phone receives a call from a contact who has special characters or accents in its name, the phone does not display the name of the contact, only the extension is shown.
• Aastra Phone Support
  • Single timezone support per TFTP server
  • Single firmware version per model supported
  • QoS: All IP packets sent by IP phones carry the same DSCP tagging, irrespective of the location of the phone.
  • When using PUM, the local directory (and speeddials) are stored on the phone itself and don't follow when logging in on a different phone
  • Speeddials and the local directory can only be provisioned via the phone itself
  • Corporate Directory search fields only in English
• Snom Phone Support
  • Only one codec supported on Snom phones

Fax limitations

• Fax server
  • The fax server only supports one outgoing identity (one single name/number can be added automatically at the top of the fax)
  • Outgoing faxes via the fax server have not been validated
• Fax devices (Template-Fax)
  • There is no fax detection, as a result a phone number can not be shared between a phone and a fax
  • Internal fax (ATA-ATA) are not supported

net.Desktop

• Supported interfaces languages: NL, FR, EN and DE
• OS support
  • supported on Microsoft Windows Vista, Windows 7
  • reported to work but not systematically tested on all OSX and Ubuntu versions
  • on Apple OSX and Ubuntu, installation is being done via Java Webstart. Java webstart is a less robust installation process compared to the Windows .msi installer.
• Contacts
  • Up to 30'000 SMP + CRM contacts can be synchronized in net.Desktop
  • Up to 100 Personal contacts can be created in net.Desktop
  • Up to 10 'Saved Queries' + 'Buddy Lists' can be created
  • In the current implementation, always all contacts are synchronized, not only the delta
• Outlook integration
  • Outlook versions 2003, 2007, 2010 are supported
  • Up to 500 Outlook contacts (public + private) can be synced towards net.Desktop
  • Click-to-dial from within Outlook is not supported except via alt-Q
• Click-to-dial via alt-q
  • not supported in Citrix, untested in other desktop virtualization solutions
  • not supported on Ubuntu and OSX
  • alt-q only works in applications where standard copy/paste is supported
  • Known issue: with X100, international numbers like +33 (0)2.565.77.77 will be transformed into 00033025657777. The redundant zero between 33 and 25 is not removed meaning it has to be removed manually.
• Video
  • Implementation based upon IAX point-to-point connection. Not SIP based, hence SIP session establishment with other SIP video clients is not supported with net.Desktop.
  • Not supported on Ubuntu and OSX
  • Supported codecs: only theora
  • Only video devices supporting the Microsoft Windows DirectShow video API are supported
• Call History
  • Call appears in history only at the end of the call
• Voicemail
  • Listening to voicemail is supported on Citrix, but other desktop virtualization technologies are untested
  • With the internal player, there is no possibility to rewind, ffwd, etc... If these features are needed, the user can play the voicemail messages via the default system player.
• Call recording
  • Not validated
  • Callflow for Supervised call recording is not included in the Template
  • SIP reinvite should be disabled in order for the RTP stream to pass through the SOP (can have impact on codec negociation)
• Identity selector: one extension per net.Desktop supported
• net.Desktop web integration
  • Web page can be openened upon an incoming call. Only the CLI can be passed as parameter.
  • Opposed to the net.Console, with net.Desktop other triggers or parameters are not supported.
  • Opposed to the net.Console, with net.Desktop the triggered web page is opened in a separate browser window.
• SMS sending and receiving not supported
• Fax: sending faxes with the embedded fax client is limited to 20 pages (no hard limit, but tested up to 20 pages)

net.Console

• All phones of a reception should be of the same type
• Only supported on Microsoft Windows Vista, Windows 7
• Not supported on OSX and Linux
• Might work but not tested under Citrix or other virtualization technologies. Citrix is under validation.
• Listening to voicemail is via IP650 (or Snom after validation test), not via internal player or system player
• Return to original receptionist via blind transfer not supported
• Active-active net.Console not supported
• A net.Console can only handle a single General and Personal queue
• Maximum 20 parkslots to park calls per SOP
• Retrieve extensions to retrieve parked calls requires extensions 700-719. These should be excluded from the customer's number range.
• Up to 40 speeddials can be defined
• Not possible to change status of receptionist or other users from within net.Console application
• No chat in net.Console
• Embedded browser
  • No proxy setting possible in embedded browser
  • Embedded browser is based upon Firefox. No specific Internet Explorer technologies supported.
• Not supported in combination with PUM phone.
• Up to 30'000 SMP + CRM contacts can be synchronized in net.Console
• One net.Console agent is linked with one physical PC and one physical phone.
• Not possible to see if internal extensions has CFU activated
• Blind transferred calls could go to second action (voicemail, forward,...) before the supervision delay (60s by default)
• When a user tries to retrieve a parked call, but the initial caller hung up in the mean time, no message will be played to announce that the caller has left, but a busy signal will be played instead